Abstract

The communication between the passengers inside a car can be difficult due to large background noise levels. It can be improved with so-called in-car communication systems. These systems capture the voice of talkers by means of microphones and play it back via loudspeakers close to the listeners. However, the challenge is the electro-acoustic feedback, which occurs when the microphone not only captures the local speech but also the loudspeaker signal. Without countermeasures, this feedback results in annoying howling sounds.

The problem of the electro-acoustic feedback has not yet been solved for in-car communication systems. Therefore, in this work techniques to suppress the feedback by means of digital signal processing are presented. The main part of this work focuses on adaptive feedback cancellation. Here, the impulse response between loudspeaker and microphone is estimated with an adaptive filter. The difficulty is a strong correlation between loudspeaker and local speech that prevents the adaptive filter from converging towards the desired solution. In order to improve convergence, a novel stepsize control is presented. As signals are not correlated during reverberation, the stepsize control exploits reverberant signal periods to update the filter coefficients. In addition to the adaptive feedback canceler, a postfilter is presented. The task of the postfilter is to suppress the residual feedback that remains after the feedback cancellation, by means of a Wiener-filter. Therefore, the postfilter is controlled depending on the adaptive filter's state of convergence. Finally, two techniques to improve the speech quality are presented. Firstly, an automatic equalizer is described that improves the sound quality. Secondly, it is shown that speech intelligibility can be improved by adding harmonics to a speech signal.

Besides the theoretical investigations, in this work also the practical realization of the algorithms is regarded. Therefore, the algorithms are integrated into a specially developed real-time framework and tested in demonstration cars under realistic conditions during numerous test drives. These test drives show a significant increase of both stability and speech quality compared to existing approaches.

Abstract

The measurement of magnetic fields for medical diagnostics is only well-established at highly specialized centers because of the high costs involved. The reason for this is the indispensable use of highly sensitive magnetic field sensors based on Super-Conducting Quantum Interference Devices. Although such systems have met the necessary technical requirements for decades, they are nonetheless expensive and very complicated to run because of cryogenic cooling. To establish the widespread use of magnetic measurements in the field of medicine, concepts for sensors that are uncooled, and thereby less expensive and user-friendly, are being researched with detection limits sufficient for measurements. A promising area of research deals with magnetoelectric sensors (ME-sensors).

To increase the usability of such sensors in realistic measurement environments and improve their signal quality with respect to the signal-to-noise ratio (SNR), this thesis examines various methods of signal processing. First, the basic procedures for measuring magnetic signals using the ME-sensors are presented. Special attention is paid to the modelling of sensor systems, the determination of the operation point, and the reduction of the signal dynamic. Due to their cantilever design, the ME-sensors have a high mechanic cross-sensitivity. Furthermore, they also measure magnetic fields of disturbing sources. To reduce their influence, the work presented here investigates different approaches based on noise cancellation. The use of a magnetic reference successfully cancels magnetic disturbances. With regard to acoustic or mechanical disturbances, various reference sensors are considered.

Irrespective of the distortion type, their influence can be reduced by up to 40 dB. Additionally, combination approaches are also investigated. These approaches are based on the idea of utilizing different frequency ranges in parallel and subsequently combining the sensor readout signals. By means of such methods, the detection limit of the sensors can be improved by more than 5 dB. In addition to this static improvement, another decisive advantage is achieved with dynamically adapting the combination. If a continuous data stream is not required and the desired signal has in principle a periodic nature, several averaging methods for an improved detection limit are discussed. In the same way, adaptive implementation of the averaging process can reduce the crosssensitivity.

These methods enabled the first biomagnetic measurement with an MEsensor by detecting the R-wave as part of a magnetocardiogram. All in all, each processing step permits continued improvement of the sensor signal with regard to their SNR. The usability of the ME-sensors in real measurement environments is thereby significantly improved.

Commission

Abstract

Speech communication inside a moving vehicle is often difficult because of the presence of high background noise levels and because the conversational partners do not face each other. In-car communication (ICC) systems help the passengers in such situations by recording the speech with microphones placed close to the talker’s mouth and reproducing it amplified with loudspeakers located close to the listener’s ears. However, by this approach, an improvement in speech intelligibility and speech quality can only be achieved if system stability, despite of the operation in a closed electro-acoustic loop, can be remained at the required system gain. Furthermore, the overall system delay has to be low enough to prevent from the perception of two individual sound sources.

Starting from the boundary conditions of speech communication inside a vehicle, this work develops a generic algorithmic framework which interconnects the signal processing methods for enhancing the microphone signals and distributing them to the available loudspeaker channels. The strict requirement for low signal delay is fulfilled by a special filter bank design which also allows for a reduction in computational complexity. In a basic version, the system stability margin is increased by equalization and signal-dependent feedback suppression. A reduction of non-stationary background noise is obtained by a multi-channel pre-processing scheme for the microphone signals. Based on this, a method for feedback cancelation is derived. Due to the high correlation between the talker signal as the desired signal and the loudspeaker signals as the excitation of the adaptive filters, a suitable method for signal decorrelation is investigated and implemented. A final comparison between different methods for feedback control clearly shows the superior performance of the cancelation approach, but also illustrates the increased requirements in system resources.

All algorithms described in this work are implemented within the real-time signal processing framework KiRAT and tested in an audio laboratory as well as under real driving conditions. Even complex algorithms, such as feedback cancelation, are always considered in the context of the entire ICC-system in order to ensure the development of practical solutions.

Abstract

A new microphone position in the automobile where microphones are placed on the seat belt is available. The pros outweigh the cons of the position which makes the it very attractive to be be practically used. In order to be able to use this microphone a set of issues are addressed through signal processing methods. Some of the methods presented are improved versions of the existing ones and some methods are new ideas. They are adapted and assessed in the automobile context. The central work of the thesis is a set of speech enhancement algorithms which are applied to the belt microphones. The speech enhancement chapters presented in the thesis form the basic units of a hands-free system.

Belt microphones when integrated into hands-free systems are used to pick up the speech of the passenger/s in the automobile. The microphone signals also contain the echoes of the remote talker which is played back over the loudspeaker in the automobile. This thesis presents an acoustic echo canceller to remove these echoes. The echo canceller must be able to not only remove the echoes to the extent that a transparent conversation is possible, but also satisfy measures specified by various standards. In order to achieve this an improved way to control the adaptive filters of the echo canceller is presented. The control involves estimation of unmeasurable quantities. Based on theoretically derived optimal quantities an improved step-size control is presented as compared to the existing ones. By utilizing properties of the impulse response such as the inherent delay, slow varying nature, it is shown that the proposed control method out performs the existing method which is based on the same principle. The chapter also presents a way to deal with the moving of the microphones when the body of the passenger moves. The problem of “room change” is handled through the two coupling factors which are in built control mechanisms of the step-size control. The step-size control and the room change is evaluated using standard measures under different realistic automobile scenarios.

The speech enhancement of the microphone also deals with the estimation of the background noise in the automobile environment. The noise estimation chapter of the thesis proposes a new noise estimation scheme applicable to the belt microphones. The scheme considers noise scenarios involving nonstationary signals like the sudden change in the noise properties when the window of the automobile is lowered. By tracking long term average, the long term level, and taking into account the short term dynamics of the background noise, a multiplicative constant based scheme is proposed. The basic idea involves the classification of the current state of the noise as either speech, slowly changing noise, or fast changing noise. Based on this classification an estimate is made combined in the end with the instantaneous background noise. By doing so it has been shown that it is possible to track the background noise aggressively at the same time avoid tracking speech. This scheme has been compared with two other well known schemes. The evaluation shows that the proposed scheme is the better choice in the evaluated scenarios.

The traditional Wiener filter approach to noise suppression has been re-looked in the final speech enhancement chapter of the thesis. The existing modifications of the Wiener filter are presented as a basis for proposing newer modifications. The first proposed modification moves from the retention of the background noise as a suppressed version to a shaping of the suppressed noise. Two ways to reshape the background noise is proposed, first involving the low frequency noise present in the belt microphones due to its proximity to the windows, second involving the equalization of the noise in the remaining frequencies. The idea is to permanently apply the low frequency modification and apply the broadband equalization to only non-speech frequencies. The second proposed modification tries to reduce the speech distortion caused when the background noise needs to overestimated to avoid the so called “musical noise”. The modifications are subjectively tested and the improvements of the methods are shown through spectrogram plots. A hands-free system where the above proposed speech enhancement algorithms has been implemented in a real-time system. This system has been tested in two cars. The software and hardware implementation details are described in the real-time implementation chapter of the thesis. The evaluation of the hands-free system into which the speech enhancement units are integrated is shown.

Commission

Prof. Dr.-Ing. Gerhard Schmidt(first reviewer)

PD. Dr. Wolfgang Koch(second reviewer)

Prof. Dr.-Ing. Michael Höft(examiner)

Prof. Dr. rer. nat. Lorenz Kienle(head of the examination board)

Abstract

An active sonar system is used for the detection, classiﬁcation and tracking of underwater objects. It consists of a transmitter which sends out acoustic energy into the water and a receiver (towing ship) which makes use of a towed hydrophone array (towed array) for detecting acoustical reflections in the environment. Within a common used signal processing chain the reflections are processed by signal enhancement and beamforming algorithms to create sonar contacts each representing one reﬂection. Based on these generated contacts, tracks of possible target trajectories are estimated in a subsequent automatic multitarget tracking which is within this work realized by a cardinalized PHD-ﬁlter. In parallel to the contact generation parameter estimation techniques could be used for approximating system parameters whose knowledge is essential for a correct contact generation. If several waveforms are used for transmission each of these waveforms is processed in an individual signal processing chain. The resulting contacts can be fused in an explicit contact fusion procedure or within the multitarget tracking to achieve an increasing localization accuracy of the targets.

A central problem in target tracking is demonstrated by the fact, that the uncertainty in the generation of sonar contacts affects the position of target tracks proportionally. The accuracy and probability of detection of sonar contacts can be mitigated by a loss of coherence in the transmitted signal waveforms and by a maneuver of the towing ship (and thus of the towed array). The purpose of this work is the study of a feedback of tracking information back to the contact generation for an improvement of the target localization. The contact fusion and the parameter estimation within the contact generation are possibilities to use the information provided by the feedback.

Within the context of this work, a so called semicoherent contact fusion is developed which counteracts the loss of coherence in the signal forms. This fusion technique contains an association of contacts generated from hyberbolic frequency modulated waveforms by using target state information from the target tracking as additional input. It is shown, that the semicoherent contact fusion leads to increasing tracking performance while suffering from multitarget situations. Within a parameter estimation for approximating the hydrophone positions the semicoherent contact fusion is used for calculating multitarget likelihoods. Within this work, the estimation of hydrophone positions is realized in a selftracking routine which also uses target state information from the target tracking. The focus of the estimation of hydrophone positions is on the moment of a towing ship maneuver. It turns out, that a self-tracking during a maneuver is possible and neither knowledge of non acoustical sensor data of the hydrophone array nor prior knowledge about sources of opportunity in the environment are necessary. The discussion of the new developed algorithms is based on Monte-Carlo simulations generating track metrics which describe the quality of the tracks at the output of the multitarget tracker. The results of the semicoherent contact fusion are compared with the iterative update procedure within the cardinalized PHD-ﬁlter. The results of the self-tracking procedure is compared with tracking results based on different knowledge about the maneuver of the array. As shown in the results, the feedback of information from the multitarget tracker back to the contact fusion and parameter estimation is a capable approach for increasing the performance of an active sonar system.

Abstract

The communication between the occupants of a speeding vehicle can be impaired. In particular, the conversation between the passengers of the front row and those of the rear seats is due to the occuring background noise and the arrangement of the passengers difficult. Usually the passengers change their position and their voice in order to compensate for this impairment. Another method to improve the communication situation is a so-called in-car communication system (ICC system). Such systems take the speech signal of a talking person using the already installed microphones, process and play back the amplified desired signal inside the passenger compartment through the loudspeakers nearby the listening passengers. Thus, the power of the desired signal inside the compartment increases as well as the intelligibility.

Following the successful implementation and development of such systems, the question arises to the quality of the ICC system. To answer this question, in this work, first, the term ”quality” referring to an ICC system is defined. Subsequently, an evaluation strategy, which consists of a variety of individual instrumental methods, is presented. For a plausible classification of the methods and a transparent investigation of the overall quality, these methods are divided into three different evaluation groups. These evaluation groups examine the characteristics of the vehicle, the behavior of the ICC system itself, as well as the quality of the communication. Within each of these groups first instrumental methods are described and validated. Subsequently, the individual results of the instrumental methods are weighted by previously calculated attributes and combined into an overall result. The attributes of the vehicle and the ICC system are determined based on the determined characteristics of the vehicle. For the development of the instrumental methods a quality reference which reflects the subjective perception of passengers is required. For this purpose, some auditory evaluation methods are introduced in addition to the instrumental methods and the results analyzed. Therefore, not only listening tests, but also psychoacoustic perception experiments have been carried out and analyzed. The evaluation strategy implemented into a real-time framework is finally tested on a specific ICC system and an overall view of the quality of all its results is proposed.

Abstract

Future development of reliable flow measurement in urban water management, flood control, process and power engineering are driven by developing wear-free and flexible measurement systems with high accuracy. Current ultrasonic measurement systems are limited in their scope and thus signal processing and sensor design need to be improved. The basis of the discussed measuring principle are sound-reflecting suspended solids in the fluid, which move slip-free along the stream.

This thesis presents fused signal processing methods to determine velocities of suspended solids in fluids. In combination with a phased-array ultrasonic sensor, a successive measurement of the velocity prole across the entire channel or pipe cross section can be performed. Moreover, signals are transmitted with a frequency hopping coding in order to detect a position-dependent information of the transit time and Doppler frequency. With the objective of noise-resistant and accurate flow measurement, object tracking theories or cluster methods are used to fuse the received echo information.

The object tracking is realized by a Kalman lter and various local and global association methods such as the Neares-Neighbour- and (Joint) Probabilistic Data Associations method. In contrast to this time-based tracking of streamlines, the cluster method determines the Doppler frequency by an application-oriented MUSIC method. Areas which are beyond of the phased-array sensor coverage, are complemented by a computational optimized function to generate a velocity distribution across the entire channel / pipe cross section. The technical feasibility of the analyzed evaluation methods as well as the sensor system has been adduced by a hardware platform.

By using the data fusion methods measurement errors have been reduced for high in flow velocities. Ambiguities in the Doppler frequency interpretation are cancelled by the data fusion, and thus enlarge the measurement range. In combination with the ultrasonic phased-array technology inhomogeneous velocity proles are determined across the entire channel / pipe cross section.

Recent News

At the end of October (24.10.2018 till 27.10.2018) we spent again a few days on the island Sylt to reflect our past, current, and future research work. This included all Ph.D. topics but also the general direction of our research. Of course, also teaching and team related issues were discussed. Beside all that official work we also enjoyed the time on that beautiful island with several common ...